...so -V1 includes the alt-preset code level tunings, but with a bitrate wetween standard and extreme.

I thought Gabriel was trying to move away from Dibrom's tuning (or "hack") by tuning in non-hack ways. In other words, hasn't he been removing more and more of Dibrom's tuning and trying to make it sound good by working from other angles? I don't understand source codes, so I can't be sure, but could someone (or Gabriel himself) comment on this? Sorry if I'm totally mistaken

I thought Gabriel was trying to move away from Dibrom's tuning (or "hack") by tuning in non-hack ways. In other words, hasn't he been removing more and more of Dibrom's tuning and trying to make it sound good by working from other angles?

No. My goal was to change the way those "tunings" are enabled. I wanted to be able to set each of those by separate parameters instead of a few "multi-purpose" parameters.

Rather than ask again - I just checked myself. --preset medium is qual=3, --preset standard is qual=3, and --preset extreme is qual=3 also according to --verbose command line when using 3.96.1. Interesting that even --preset insane is qual=3. It appears that -q 0/1/2 isn't used in any of the presets, unless I have something in EAC set wrong and it is adding other command line switches I don't know about.

Rather than ask again - I just checked myself. --preset medium is qual=3, --preset standard is qual=3, and --preset extreme is qual=3 also according to --verbose command line when using 3.96.1. Interesting that even --preset insane is qual=3. It appears that -q 0/1/2 isn't used in any of the presets, unless I have something in EAC set wrong and it is adding other command line switches I don't know about.

What's with the q-value obsession? I'm not picking on you detokaal, I've been seeing it creeping up in other discussions. AFAIK, the -V settings were tuned using a given q-value and changing this isn't a reliable way to adjust quality. Rather just change the -V value. It seems the -V values have been distributed to cover the full bitrate spectrum, so it seems that anyone could find the ideal quality/space tradeoff for their particular ears and application.

Though I guess if you really want to be experimental, go for it. Might as well start tweaking the code as well. Perhaps Gabriel can either validate or invalidate this point.

For ~ 128kbps VBR, the general consensus has been to use "-V5 --athaa-sensitivity 1" with LAME 3.95.1 or higher. The ATH adjustment was found to result in less warbling/phasing, but may raise the bitrate a little bit.

I just registered myself for the first time ever to a forum of anykind

But this just seemed like a place of people who know the stuff they talk about,and that's always good. Also this place has had by far the best attitude towards"the new guy" making his first post, so bear with me, since this truly is my first post

So, here's my thing, I'm in a situation now to encode all of my audio cds to mp3 format.

By judging form the discussions here, i really should use eac or plextoolsfor the ripping and lame (3.90.3 or 3.96.1 if i'm correct?) for the encoding.

Even more reading of this forum has led me to the conclusion to use vbr ape or as you would put it in the format of -V0 while using lame 3.96.1.

So the thing I would like to know is about the q value,is there any point in adjusting it upwards trough the commandline myself?

I mean seriously, I thought about this long time, that do I even dare to ask such a question,because I respect what Gabriel has stated about the defaults,but I just want to know that what are the potential effects of changingthe qval to 2 or even 1 or 0. What does it really change?Does it affect the quality of the audio in any way?I even read about it potentially lessening the outcoming quality of the finished "product",is that still true?

well nothing more at this point, hope this didn't strike you guys as a totally stupid question.

use LAME 3.96.1 and -V5 --athaa-sensitivity 1. It results better qualits than -V5 (applies to -V5 only!) I've encoded 5 files or so and in average they were 128kbps. However, it is possible that some files have 111kbps and others 134kbps...Even if you get a bit over 128kbps in average, let's say 132kbps...who cares...it's not that much of a difference (in my example 0,5 kilobyte per second = 30 kilobyte per minute)...not a big deal, but the quality is the best mp3 has to offer in the 128kbps range...

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--alt-presets are there for a reason! These other switches DO NOT work better than it, trust me on this.LAME + Joint Stereo doesn't destroy 'Stereo'

use LAME 3.96.1 and -V5 --athaa-sensitivity 1. It results better qualits than -V5 (applies to -V5 only!) I've encoded 5 files or so and in average they were 128kbps. However, it is possible that some files have 111kbps and others 134kbps...Even if you get a bit over 128kbps in average, let's say 132kbps...who cares...it's not that much of a difference (in my example 0,5 kilobyte per second = 30 kilobyte per minute)...not a big deal, but the quality is the best mp3 has to offer in the 128kbps range...

Actually, the --athaa-sensitivity 1 switch was also recommended for -V4 on some thread I can't seem to find right now. I use it for my -V4 encodes and I haven't noticed any problems at all. (Actually, I can't ABX the difference, but if the golden ears say it's better, I trust them )

I just registered myself for the first time ever to a forum of anykind

But this just seemed like a place of people who know the stuff they talk about,and that's always good. Also this place has had by far the best attitude towards"the new guy" making his first post, so bear with me, since this truly is my first post

So, here's my thing, I'm in a situation now to encode all of my audio cds to mp3 format.

By judging form the discussions here, i really should use eac or plextoolsfor the ripping and lame (3.90.3 or 3.96.1 if i'm correct?) for the encoding.

Even more reading of this forum has led me to the conclusion to use vbr ape or as you would put it in the format of -V0 while using lame 3.96.1.

So the thing I would like to know is about the q value,is there any point in adjusting it upwards trough the commandline myself?

I mean seriously, I thought about this long time, that do I even dare to ask such a question,because I respect what Gabriel has stated about the defaults,but I just want to know that what are the potential effects of changingthe qval to 2 or even 1 or 0. What does it really change?Does it affect the quality of the audio in any way?I even read about it potentially lessening the outcoming quality of the finished "product",is that still true?

well nothing more at this point, hope this didn't strike you guys as a totally stupid question.

thanks.

That is a valid question, one I bothered people with myself. -q 0 increases the quality of the psychoacoustic algorithm, it does not touch the bitrate at all.In simple terms Lame will spend much more time examining each sample when encoding.

Lame 3.90.3 at --alt-preset standard is the recommended mp3 settings. Lame 3.97 final might replace 3.90.3 as the best encoder.

I use lame 3.96.1 -V 5 --athaa-sensitivity 1 -q 0, which might not improve the quality much over not using -q, but encodes fast enough for me and sounds good on my apex dvd player and my pc/stereo setup.

For cd's that I have to archive and won't have access to in the future, I use Musepack/MPC at --standard. MPC has fewer problem samples at higher bitrates and is tuned only for high transparent bitrates, but is not supported by any portables . It is very good for computer use.

Lame 3.90.3 at --alt-preset standard is the absolute best mp3, except in certain samples.

preset standard is not the "absolute best" mp3. It's a compromize between transparency (I'd rather say 'robustness') and bitrate. For many people, --preset standard is not a good compromize: they could obtain the same level of transparency at lower bitrate with lower presets. For other people, --preset extreme improves the quality, and not only on killer samples.For absolute best mp3, use --preset insane (or even freeformat at 640 kbps).